The Internet has evolved into an essential communication tool for millions of users in the business, technical and educational fields. In this regard, a growing use of the Internet relates to Internet telephony which provides a number of advantages over conventional circuit-switched network controlled by a separate signaling network.
An important feature in most modern telephony systems is voice messaging. An extension of voice messaging is unified messaging, where access to messaging services in various media is provided in a common platform. For example, the ability to converge voice mail, e-mail, video messaging, instant messaging services and the like within a common system as part of the telephony network provides a single platform for users to conveniently access such services.
Voicemail service is generally provided by the local private branch exchange (PBX) or local exchange carrier. Such current voice mail systems are typically closed architectures. As a result, it is often difficult to perform simple operations, such as forwarding voicemail outside the local PBX, filtering or sorting of messages. Thus, an open architecture which facilitates simple data exchange within and without of the local telephony exchange would be desirable.
The session initiation protocol (SIP) is gaining in popularity as a standard signaling protocol for use in Internet telephony. As this popularity grows, it would be desirable to provide a system architecture and method for providing unified messaging services on a SIP based system. In addition, the real time streaming protocol (RTSP) has been proposed as a standard transport protocol for multimedia service, such as video, audio and mixed media files, over the Internet. A unified messaging system which employs SIP as the signaling protocol along with RTSP for message storage and delivery can offer many benefits over known messaging systems.